The present invention relates to telecommunication systems in general, and in particular to the transmission of compressed signals in telecommunications systems.
In recent years, various techniques are being implemented in order to save on required bandwidth, techniques which achieve toll-quality or near toll-quality speech while using compressed telecommunication transmissions. These techniques typically involve the use of coding algorithms that allow reducing the bandwidth requirement of 64 kb/s for non-compressed transmissions. One such example is the LD-CELP algorithm that enables reducing the bandwidth requirement to 16 kb/s. Naturally, in order to use such coding algorithms, both ends of the transmission path must be provided with the ability to code and decode the transmissions. One solution for this requirement is using single proprietary equipment at both ends of and along the transmission path. Another possible solution is the implementation of international standards that allow compatibility of different types of equipment located along a transmission path.
The international standard for the coding algorithm LD-CELP was published on March 1995 as International Telecommunication Union (ITU-T) Recommendation G.728. However, it was found that this Recommendation contained several drawbacks. Among these drawbacks was the handling of transmissions at variable bit rate (referred to hereinafter as xe2x80x9cVBRxe2x80x9d). This problem was particularly noticed when G.728 Recommendation was used in voiceband data applications.
In its contribution to the ITU-T of Mar. 17, 1997, ECI Telecom Ltd. suggested a solution disclosed in Annex J of ITU-T Recommendation G. 728. The contribution, entitled xe2x80x9cVariable Bit-Rate algorithm, mainly for the Voiceband data applications of LD-CELP ITU-T Rec. G. 728 in DCMExe2x80x9d is hereby incorporated by reference. This publication will be referred to hereinafter as xe2x80x9c40 kbps algorithmxe2x80x9d.
In this contribution, a solution for VBR and particularly for voice-band data (to be referred to hereinafter as xe2x80x9cVBDxe2x80x9d) application, was described. The contribution provided information for the implementation of a codec that complies with the LD-CELP algorithm, as well as modification to Annex G of Rec. G 728, xe2x80x9c16 kbit/s fixed point specificationxe2x80x9d, so as to enable a mode-switch on a fixed point arithmetic device.
The codec described in the 40 kbps algorithm basically uses a transmission rate of 40 kbit/s. The algorithmic delay is 5-samples long, totaling 0.625 msec, and the codec can perform a mode-switch every xe2x80x9cadaptation-cyclexe2x80x9d (2.5 msec).
The suggested 40 kbps algorithm, was intended mainly to solve problems in the transmission of compressed VBD for applications such as DCME, and was suggested to replace the 40 kbps ADPCM mode (ITU-T Rec.xe2x80x94G.726) in DCME systems where LD-CELP algorithm is incorporated. Among the features provided by this algorithm is the soft transition to and from the LD-CELP algorithm, and the maintaining of toll-quality or near toll-quality of speech.
The adaptation cycle used for the speech mode in the 40 kbps algorithm is essentially provided by G. 728 Recommendation. Therefore, when reverting to speech mode type of operation, the LD-CELP mode specified in Recommendation G.728 will be applied rather than the 40 kbps algorithm.
The main modification of a codec operating in accordance with the 40 kpbs algorithm is the implementation of the Trellis Coded Quantization (referred to hereinafter as xe2x80x9cTCQxe2x80x9d) approach, described in IEEE Transactions on Communications Vol. 38, No. 1, (1990) which is hereby incorporated by reference. This TCQ approach replaces the analysis-by-synthesis approach to codebook search of ITU-T Rec. G. 728, in the VBD mode.
Still, in the 40 kbps algorithm suggested, no solution was provided to the problem of how to avoid reaching a saturation state when an impulse occurs in the prediction error, e.g. when having a sudden substantial change in the energy level of the prediction error. This problem results in generating a high level of noise at the output of the decoder, and is known to be a cause for discrepancies between the transmitting and the receiving ends of the transmission path.
U.S. Pat. No. 4,677,423 recognizes a somewhat similar problem associated with another type of algorithm, the ADPCM algorithm, and discloses a solution to that problem. The mechanism described in U.S. Pat. No. 4,677,423 is one for overcoming the problem associated with transitions in partial band energy signals, by locking and unlocking the adaptation speed. The adaptation speed is locked in cases of very slow speed of adaptation, while the unlocked mode is used when high speed of adaptation is required. Unfortunately, since this solution is not fast enough for systems having coding algorithms where the predictor is not an adaptive one, e.g. based on Linear Prediction (referred to hereinafter as LP) analysis, a different solution is required. A number of problems render the solution described in U.S. Pat. No. 4,677,423 inefficient when trying to avoid saturation in systems incorporating linear predictors, when an impulse occurs in the prediction error. Some of these problems are: the ""423 solution is based on fact that each sample should be handled individually, whereas in linear predictors, a vector comprising a number of samples is used rather than single samples as suggested in the ""423 solution, a difference which renders the ""423 solution not fast enough to be applied in linear predictors systems. Another basic difference is, that the-errors handles by the ""423 patent are logarithmic errors which are not likely to saturate the quantizer as fast as linear errors might. Therefore a different solution is required, one that can provide an answer to systems where linear predictors are incorporated.
It is therefore an object of the present invention to provide a method for determining the compensated scaling of a quantizer in a coder using a vectorial linear non-adaptive predicting algorithm, a method that overcomes the drawbacks of the prior art solutions described above.
It is another object of the present invention to provide a digital communication apparatus and system enabling to overcome problems caused by impulses occurring in the prediction error.
Further objects and features of the invention will become apparent from the following description and the accompanying drawings.
In accordance with the present invention there is provided a method for determining the compensated scaling of a quantizer in a process of encoding/decoding a VBD type transmission by using a vectorial linear non-adaptive predicting type algorithm.
The term xe2x80x9cVBDxe2x80x9d as will be referred hereinafter, is used to denote digital signals modulated for transmission in the voice band frequency (up to 4 KHz), e.g. modem signals, DTMF signals, or any other such narrow band type of signals.
The method provided by the present invention, preferably comprises the steps of:
i. providing a digital sample vector in a coded form;
ii. calculating LP coefficients for predicting said digital sample vector and deriving a linear prediction error vector therefrom;
iii. calculating the gain of said linear prediction error vector;
iv. calculating the scaling of the quantizer from said gain;
v. calculating an average value of said gain corresponding to said digital sample vector, based on preceding digital samples;
vi. calculating the difference between said gain and said average value;
vii. determining whether a gain compensation is required for an impulse in the prediction error of said digital sample vector, based on:
(a) comparing said difference with a first pre-defined threshold value, and
(b) comparing the differences between the gains associated with a pre-defined number of most recent digital sample vector provided and their corresponding average values and a second pre-defined threshold;
viii. in the case that the determination in step (vii) is that a gain compensation is required, determining the compensation required for the impulse in the prediction error of said digital sample vector;
ix. combining the scaling of the quantizer as obtained by step (v) with the gain compensation determined in step (viii) to obtain the compensated scaling of the quantizer.
An example of such a linear non-adaptive predicting algorithm is an algorithm of the type all poles modeling.
The determination whether a signal can be qualified as a steady signal, is done by comparing the differences existing between the gains associated with a pre-defined number of preceding digital sample vectors and the average values associated therewith, with the second pre-defined threshold. If these differences do not exceed that second pre-defined threshold, the signal may be qualified as a steady signal.
According to a preferred embodiment of the invention, the method described further comprises a step of calculating the value of a pre-defined function, which function is based on the calculated LP coefficients associated with the digital sample vector. The value of the pre-defined function thus obtained may be used in determining the required gain compensation. According to this embodiment, this can be done for example, by setting a constrain that unless the calculated value is higher than that of a pre-defined value, no gain compensation will be carried out. Another possible example is by applying a factor on the gain compensation that depends on the difference existing between the calculated value and that of the pre-defined value.
An example of such a pre-defined function according to this embodiment is a function that is equal to       ABS    ⁡          (              A        ⁡                  (          1          )                    )                  ∑              i        =        1            11        ⁢          xe2x80x83        ⁢          ABS      ⁡              (                  A          ⁡                      [            i            ]                          )            
where A[i] are the LP coefficients.
Similarly, as can be appreciated by any person skilled in the art, other gain compensation decision mechanisms can also be used and their results be incorporated in the final decision upon the actual compensation to be carried out.
According to yet another embodiment of the present invention, a peak threshold value is pre-defined, and the calculated value of the difference as calculated in step (v) of the above method, is compared with that peak threshold. This embodiment enables among others, extending a first pre-defined period of time during which the gain is compensated while its value does not exceed that peak threshold. The gain compensation period can be extended for example until either the peak is reduced below the level of that peak threshold, or to a longer, pre-defined period of time.
According to still another preferred embodiment of the present invention, the linear prediction error vector is derived by performing a Trellis code quantization on the prediction error vector, and selecting a preferred quantized linear prediction error vector from among a number of quantized linear prediction error vectors calculated. More preferably, such selection is made by choosing the linear prediction error vector that has the minimal prediction error.
According to a further embodiment of the present invention, the determination of the gain compensation required as set at step (viii) is subjected to a limiting threshold to prevent from reaching over-compensation of the gain.
By another aspect of the present invention, there is provided digital telecommunication station operative in a digital telecommunication system, and comprises:
input interface adapted to receive a voiceband data signal and operate thereon;
processing means adapted to calculate:
LP coefficients for predicting said digital sample vector and deriving a linear prediction error vector therefrom;
the gain of said linear prediction error vector;
the scaling of the quantizer based on said gain;
an average value of said gain corresponding to said digital sample vector, based on preceding digital samples;
the difference between said gain and said average value;
first determination means for determining whether a gain compensation for the impulse in the prediction error of said digital sample vector is required, based on:
a. comparing said difference with a first pre-defined threshold value, and
b. comparing the differences between the gains associated with a pre-defined number of most recent digital sample vectors save that of said digital sample vector provided and their corresponding average values and a second pre-defined threshold,
second determination means adapted to determine the gain compensation required to compensate for the impulse in the prediction error of said digital sample vector if the determination made by the first determination means is affirmative;
means for combining the scaling of the quantizer with the gain compensation determined by said second determination means; and
output interface adapted to transmit a voiceband data signal.
As would be appreciated by a person skilled in the art, the device described above may comprise further features that are known in the art per se, and should thus be understood as being encompassed by the present invention.
The term xe2x80x9ctelecommunication networkxe2x80x9d, as will be used hereinafter, should be understood to encompass the various types of networks known in the art, such as TDM, synchronous and asynchronous transfer networks, IP networks, IP frame relaying networks and any other applicable communication networks.
The term xe2x80x9ctelecommunication stationxe2x80x9d is used herein to describe a combination of at least one pair of encoding/decoding devices, one of which is used for converting, when required, signals received to a new coded form, while the other is used as its corresponding decoder, converting signals received in this new coded form to essentially their pre-encoder form. Such two devices may either be included within one apparatus or be separated from each other.
According to still a further embodiment of the invention there is provided a telecommunication apparatus operative in a digital telecommunication system and adapted to produce temporal change in quantization gain in a process of encoding/decoding transmission of the VBD type, comprising the following:
i. gain average calculator;
ii. impulse detector;
iii. signal classifier;
iv. decision means; and
V. gain compensator.
According to another preferred embodiment, the average calculator is operative to calculate the average of the gain estimation by using the most recent vector gain value, and the difference, Gdiff, between said most recent vector gain value and said average of the gain compensation. More preferably, the difference Gdiff is received and compared with a pre-determined first threshold, by the impulse detector which is operative to detect sudden changes in the gain after a predetermined period of time.
According to yet another preferred embodiment of the present invention the signal classifier is adapted to detect pre-defined VBD transmissions, and more preferably, the decision means is adapted to receive the output of the impulse detector and the signal classifier, and to activate the gain compensator accordingly.
By still another preferred embodiment, the gain compensator is operative to increase the gain for a pre-defined period of time.
According to another aspect of the invention there is provided a digital communication system for interconnecting a plurality of telecommunication trunks via a transmission path, comprising:
first transmission means at at least a first end of the transmission network for transmitting digital signals;
at least one pair of telecommunication stations of the type specified, and
receiving means at at least a second end of the transmission network.